VoIP system and method for preventing data loss in the same

ABSTRACT

The present invention relates to a VoIP (Voice over Internet Protocol) system and a method for preventing data loss in the same. A receiving system in the VoIP system decides a hold time corresponding to the loss rate of compressed voice data, and provides the hold time to a transmission system. The transmission system adjusts voice input speed depending on the hold time. Therefore, the transmission system can regulate compressed voice data generation rate therein not to exceed a designated channel capacity.

BACKGROUND OF THE INVENTION

[0001] 1. Field of the Invention

[0002] The present invention relates to a VoIP (Voice over InternetProtocol) system and a method for preventing data loss in the same.

[0003] 2. Description of the Related Art

[0004] Most current systems for voice over an Internet Protocol (VoIP)are built as described in a block diagram (FIG. 1) which illustrates aconstruction of the VoIP system for voice communication, that is,placing and receiving internet-based calls in prior art.

[0005] As illustrated in FIG. 1, basically the conventional VoIP systemfor voice communication through internet comprises a transmission system50 that converts the recipient's voice to voice data and transmits thevoice data over the internet, and a receiving system 60 that receivesthe voice data from the transmission system and converts the voice databack to the voice. The transmission system 50 is equipped with amicrophone 10, ADC (Analog to Digital Converter) 11, a voice encoder 12,and a transmitting protocol processing part 13. The receiving system 60is equipped with a receiving protocol processing part 14, a voicedecoder 15, DAC (Digital to Analog Converter) 16, and a speaker 17.

[0006] The conventional VoIP system is operated as follows:

[0007] The microphone 10 in the transmission system 50 creates ananalogue voice signal from the inputted voice by the transmitter, andthe ADC 11 converts the analogue voice signal to a digital voice signal.

[0008] The voice encoder 12 compresses the digital voice signal togenerate compressed voice data, and the transmitting protocol processingpart 13, in order to transmit the compressed voice data to the receivingpart over the internet, attaches a header and a trailer to thecompressed voice data and generates voice packets.

[0009] On the other hand, the receiving protocol processing part 14 inthe receiving system 60 analyzes the voice packets received from thetransmission part, and extracts the compressed voice data from the voicepackets by removing the header and the trailer that are installed in thevoice packets.

[0010] The voice decoder 15, through decompression of the compressedvoice data, restores the digital voice signal, and the DAC 16 convertsthe digital voice signal to an analog voice signal.

[0011] The speaker 17 converts the analog voice signal to the voice ofthe transmitter to help the recipient to be able to listen to thetransmitter's voice.

[0012] As explained above, voice communication quality over theVoIP-based system is generally dependent on the number of voice packetsper time unit (i.e., compressed voice data generation rate)corresponding to the conversation speed of the transmitter as inputtedthrough the microphone 10 in the transmission system 10.

[0013] In other words, in case the inputted conversation speed exceedsthe allowable number of voice packets for transmission per time unit(i.e., Channel Capacity) over the internet, some of packetscorresponding to the difference from the compression voice datageneration rate and the channel capacity are not transmitted to therecipient and get lost, consequently deteriorating the speech quality.

[0014] Therefore, the transmitter is encouraged to adjust his or herconversation speed to be inputted through the microphone 10 in order toensure that the voice packets he or she transmitted are safelytransmitted to the recipient without a loss.

[0015] For example, since the compression voice data generation rateshould not exceed the channel capacity assigned to himself or herself,the transmitter can temporarily stop voice input for a certain period oftime and continue later for decreasing the compression voice datageneration rate, in case the compression voice data generation rateexceeds the designated channel capacity to himself or herself.

[0016] Previously, the recipient, when there was a blank while listeningto the transmitter's voice, let the transmitter know that he or she washaving difficulty in catching the transmitter's voice, and thetransmitter adjusted his or her voice volume accordingly.

[0017] However, the previous method described above had a problembecause the conversation speed control was made primarily by thetransmitter's sense, making it difficult to suppress the compressedvoice data generation rate under the channel capacity. Thus, it was veryhard to prevent the voice packet loss in the conventional system.

SUMMARY OF THE INVENTION

[0018] It is, therefore, an object of the present invention to provide aVoIP system in which a receiving system determines a hold time accordingto compressed voice data loss rate and a transmission system adjusts theconversation speed using the hold time, and a method for preventing dataloss in the VoIP system.

[0019] Another object of the present invention is to provide a VoIPsystem that regulates the compressed voice data generation rate of atransmission system not to exceed channel capacity, and a method forpreventing data loss in the VoIP system.

[0020] To achieve the above object, there is provided an method forvoice communication over the above VoIP system in which the receivingsystem calculates the loss rate of the compressed voice data from thetransmission system, and based on this calculation, determines a holdtime for intercepting any further voice input. Then, the hold time isprovided to the transmission system.

[0021] According to the preferred embodiment of the present invention,the VoIP system is equipped with a transmission system and a receivingsystem.

[0022] In the receiving system, there are two kinds of module: one is avoice data loss detection module to calculate the loss rate of thecompressed voice data from the receiving system, and the other is a holdtime decision-making module that determines the current hold time on thebasis of the compressed voice data loss rate from the voice data lossdetection module, generates a hold time information packet whichincludes an information on the current hold time, and transmits thegenerated hold time packet to the transmission system over the internet.

[0023] When the hold time information packet from the hold timedecision-making part is received, the hold time display part in thetransmission system displays the hold time to the transmitter.

[0024] Another embodiment of the present invention provides a method forpreventing data loss in the VoIP system equipped with a transmission andreceiving system, the method comprising the steps of: extractingcompressed voice data from voice packets that are transmitted by thetransmission system, calculating loss rate of the voice data based onthe extracted compressed voice data, deciding a hold time forintercepting the transmitter's voice input to decrease the loss ratedown to an allowable limit, and transmitting the hold time informationto the transmission system.

[0025] Therefore, the transmitter temporarily stops his or her voiceinput during the hold time so that the compressed voice data loss can beprevented, and the VoIP system can perform better quality of voicecommunication.

BRIEF DESCRIPTION OF THE DRAWINGS

[0026] The above objects, features and advantages of the presentinvention will become more apparent from the following detaileddescription when taken in conjunction with the accompanying drawings, inwhich:

[0027]FIG. 1 is a block diagram illustrating a construction of a VoIPsystem voice communication over the internet according to theconventional VoIP system;

[0028]FIG. 2 is a block diagram illustrating a construction of the VoIPsystem according to the preferred embodiment of the present invention;

[0029]FIG. 3 is a flow chart illustrating the procedure of determining ahold time and transmitting the hold time by the VoIP system according tothe preferred embodiment of the present invention;

[0030]FIG. 4 is a diagram illustrating an information packet format forthe hold time;

[0031]FIG. 5 is a flow chart with more details than FIG. 3.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT

[0032] A preferred embodiment of the present invention will now bedescribed with reference to the accompanying drawings. In the followingdescription, same drawing reference numerals are used for the sameelements even in different drawings. The matters defined in thedescription such as a detailed construction and elements of a circuitare nothing but the ones provided to assist in a comprehensiveunderstanding of the invention. Thus, it is apparent that the presentinvention can be carried out without those defined matters. Also,well-known functions or constructions are not described in detail sincethey would obscure the invention in unnecessary detail.

[0033]FIG. 2 is a block diagram illustrating a construction of the VoIPsystem according to the preferred embodiment of the present invention.

[0034] As illustrated in FIG. 2, the receiving system determines a timefor intercepting voice input by the transmission system (hereinafter, itis referred ‘hold time’), which is essential for regulating compressedvoice data generation rate not to exceed channel capacity, and the holdtime is transmitted to the transmission system over the internet. Thetransmission system, therefore, stops the voice input during the holdtime.

[0035] The major components of the receiving system are a voice dataloss detection part and a hold time decision-making part, in which thefirst calculates a compressed voice data loss rate and the latterdetermines a hold time based on the voice data loss rate from the voicedata loss rate part, generates hold time information packets includingthe hold time information, and transmits the generated packets to thetransmission system over the internet.

[0036] Meanwhile, the most important component in the transmissionsystem is a hold time display part that receives the hold timeinformation packets from the hold time decision-making part, anddisplays the hold time corresponding to the hold time informationpackets to the transmitter.

[0037] Thus, the transmitter can temporarily stop his or her voice inputduring the hold time, and in result, the VoIP system can prevent dataloss due to excessive data therein.

[0038] The constructions of the transmission system and the receivingsystem of the VoIP system are explained in more detail with reference toFIG. 2.

[0039] As illustrated in FIG. 2, the transmission system 100 consists ofa microphone 10, ACD 11, a voice encoder 12, a transmitting protocolprocessing part 13, and a hold time information display mole 110.

[0040] On the other hand, the receiving system 200 consists of areceiving protocol processing part 14, a voice decoder 15, DAC 16, aspeaker 17, a voice data loss detecting part 210, and a hold timedecision-making part 220.

[0041] The transmission system 100 converts the transmitter's voiceinputted through the microphone 10 to generate voice packets therefrom,and transmits these voice packets to the receiving system 200 over theinternet. The receiving system 200 processes the voice packets from thetransmission system 100 to generate an analog voice signal of thetransmitter, and later, the analog voice signal is output as the abovevoice through the speaker 17 for the recipient.

[0042] The following explains the operation of the VoIP communicationsystem shown in FIG. 2. For the components of FIG. 2, since they are thesame construction with those of FIG. 1, and have the equivalentcapabilities, the same numerical references as those of FIG. 1 are used,and the details in those components are accordingly omitted here.

[0043] As shown in FIG. 2, the receiving system 200 includes a hold timeprocessing part 250.

[0044] The hold time processing part 250 consists of the receivingprotocol processing part 14, the voice data loss detection part 210, andthe hold time decision-making part. The transmission system 100 isequipped with the hold time information display part 110 correspondingto the hold time processing part 200.

[0045] How the hold time processing part 250 works can be explained thatthe protocol processing part 14 in the receiving system 200 extracts thecompressed voice data from the voice packets received over the internet,and transmits the extracted compressed voice data to the voice decoder15 and the voice data loss detection part 210, respectively.

[0046] At this point, the compressed voice data is generated in thetransmission system where the compressed voice data is converted to apacket format, and the transmission system transmits the voice packet tothe receiving system 200.

[0047] The voice data loss detection part 210 calculates the compressedvoice data loss rate using the compressed voice data received from thereceiving system, and transmits the loss rate to the hold timedecision-making part 220.

[0048] The hold time decision-making part 220 receives the compressedvoice data loss rate, and determines a hold time based on this loss rateso that the compressed voice data generation rate does not exceed thedesignated channel capacity.

[0049] The hold time decision-making part 220 then transmits the holdtime to the hold time information display part 110 over the internet.The hold time information display part 110, using the hold timeinformation, informs the transmitter of the hold time through a screenthat is installed in a terminal of the transmission system 100 or LED(light emitting diode) and so forth.

[0050]FIG. 3 is a flow chart explaining the procedure from deciding ahold time to transmitting the hold time in the VoIP system according tothe preferred embodiment of the present invention.

[0051] As shown in FIG. 3, the protocol processing part 14 of thereceiving system 200 receives voice packets that are transmitted fromthe transmission system 100 over the internet, and extracts thecompression voice data from the voice packets. This extracted compressedvoice data is then transmitted to the voice data loss detection part210.

[0052] The detection of the voice data packet loss is proceeded asfollows.

[0053] First of all, each voice data packet has a RTP protocol header,and the number of sequence per packet is installed in the header. Thus,while receiving the voice data packets, if the sequence number installedin the header increases more than 2 units, not 1 unit in sequence asusual, the voice data loss detection part 210 detects that there is aloss on the voice data packet as much as the increment.

[0054] That is, the voice data loss detection part 210, after itreceives the compressed voice data, figures out the number of lost voicepackets by reading the packet order of the header in the voice packets(S31).

[0055] The voice data loss detection part 210 then compares the numberof the voice packets transmitted from the transmission system 100 withthe number of the lost voice packets, and calculates the packet a lossrate of the compressed voice data (S32).

[0056] The method for calculating a packet loss rate of the compressedvoice data is now explained.

[0057] First of all, the on-chip memory records the number of the voicedata packet D that is received from the voice data loss detection part210 every 30 seconds.

[0058] As described above, each voice data packet has the RTP protocolheader information and the packet sequence number. Thus, with thesenumbers, the voice data loss detection part 210, calculates the totalvoice data packet number (T) to be received from the maximum number (M)of the sequence and the minimum number (N) of the sequence among thevoice data packets that are received for 30 seconds. The equation (1)therefor is as follows:

T=M−N - - -   (1)

[0059] Further, the voice data loss detection part 210 calculates theloss rate(L) of the voice data packets as explained in the followingequation (2):

L=(T−D)/T - - -   (2)

[0060] For instance, the transmission system 100 transmitted 100 ofvoice packets for a designate time unit, but when taken for theanalysis, the receiving system 200 only received 90 voice packets out ofthe total 100 voice packets, losing the remaining 10 voice packet, thecompressed voice data packet loss rate in this case is 10%, that is,(10/100)*100=10%.

[0061] Later, the voice data loss detection part 210 transmits thecalculated packet loss rate(L) of the compressed voice data to the holdtime decision-making part 220.

[0062] In the meantime, the hold time decision-making part 220 decidesthe hold time S33 by comparing the voice data loss rate with anallowable loss rate (hereinafter, it is referred “allowable value”).

[0063] The following explains how the hold time decision-making part 220determines the hold time.

[0064] The voice data packet loss rate (L) is calculated every 30seconds. Here, the initial value of the hold time is 0 second. Bycalculating the loss rate (L) per 30 seconds, if the loss rate (L) ishigher than 5%, the hold time decision-making part 220 extends the holdtime by 1 second. However, the hold time should not be longer than 5seconds. Similarly, if the loss rate (L) per 30 seconds is less than 1%,the hold time decision-making part 220 decreases the hold time by 1second. However, in this case, the hold time should not be shorter than0 second. In the meantime, if the loss rate (L) is between 1% and 5%,the previous hold time continues as the current hold time.

[0065] Next, the hold time decision-making part 220 generates hold timeinformation packets using the hold time information obtained from theabove.

[0066]FIG. 4 illustrates a format of the hold time information packet.

[0067] A in the FIG. 4 is a service identifier and indicates the holdtime information packet. In the invention example, the A is 4 bytes,0000 0000 0000 0000 0000 0000 0000 0000.

[0068] B in the FIG. 4 is a session ID number, which is given to everytelephone call. In the invention example, the B is 3 bytes.

[0069] C in the FIG. 4 indicates a hold time, which means the waitingtime before starting the next sentence during a telephone call. In theinvention example, C is 1 byte. If the C is 0000 0000, 0000 0001, 00000010, 0000 0100, 0000 1000, and 0001 0000, the corresponding hold timeis 0 second, 1 second, 2 seconds, 3 seconds, 4 second, and 5 seconds insequence.

[0070] The hold time decision-making part 220 transmits the voice datapackets formatted as FIG. 4 to the hold time information display part110 of the receiving system 200 over the internet (S34).

[0071] On the other hand, when the hold time information packet isreceived, the hold time information display part 110 installed in thereceiving system 200 displays the hold time to the transmitter throughLED (light emitting diode) or the monitor (S35).

[0072] Therefore, the transmitter can control the conversation speed bytemporarily stopping his or her voice input during the hold time inorder to maintain the compressed voice data generation rate below thedesignated channel capacity.

[0073]FIG. 5 is a detailed flow chart illustrating the procedure ofdecision-making the hold time as in FIG. 3, and the procedure ofgeneration and transmission S34 of the hold time information packet.

[0074] To begin with, the hold time decision-making part 220 receivesthe compressed voice data packet loss rate (L) from the voice data lossdecision-making part 210. Then, the hold time decision-making part 220decides whether the compressed voice data packet loss rate (L) exceedsthe designated allowable value or not (S42).

[0075] In the invention example, the designated allowable values are 5%and 1%. If the loss rate (L) reaches 5%, the voice corresponding to 0.5second out of 10 seconds is lost, which creates difficulty for the userduring a telephone call since the lost voice corresponding to the 0.5second is the same with the length of one syllable. Meantime, if theloss rate (L) becomes 1%, the voice corresponding to 0.1 second out of10 seconds is lost, which makes no difference to the user. In this case,the lost voice corresponding to the 0.1 second is much shorter than thelength of one syllable.

[0076] Meanwhile, the voice encoder 12 in the transmission system 100uses international standard voice compression, G.723.1 or G.729A.

[0077] At the result of decision, if the compressed voice data packetloss rate (L) is higher than the allowable value, i.e., 5%, the holdtime decision-making part 220 increases the previous hold time (here,the initial value of the hold time is ‘0’) by 1 second (S46). At thistime, if the hold time is too long, the communication between thetransmitter and the recipient might not be in the good condition, thusit becomes necessary to predesignate the maximum hold time (e.g., in theinvention example, it is 5 seconds.). Thus, it is evaluated whether ornot the increased hold time is larger than the maximum hold time(hereinafter, it is referred to ‘maximum value’) (S47).

[0078] If the increased hold time is larger than the maximum value, themaximum value is re-adjusted to be equal to the current hold time (S48).However, if the increased hold time is not larger than the maximumvalue, the increased hold time is set to be the current hold time.

[0079] On the other hand, if the compressed voice data loss rate (L) isnot larger than the allowable value (e.g., 1% as in the inventionexample), the hold time decision-making part 220 decreases the previoushold time by 1 second (S43). Since the hold time can not have a negativevalue, the reduced hold time can't be less than ‘0’.

[0080] Further, the hold time decision-making part 220 determineswhether or not the reduced hold time is lower than ‘0’. Then, if thereduced hold time is less than ‘0’, the hold time decision-making part220 designates ‘0’ as the current hold time (S45). However, if thereduced hold time is not less than ‘0’, the reduced hold time is set tobe the current hold time.

[0081] As explained before, the hold time decision-making part 220generates the hold time information packet including the current holdtime information once the current hold time is determined by increasingor decreasing the previous hold time after comparing the compressedvoice data loss rate and the allowable values (S49).

[0082] The hold time information packets are transmitted over theinternet to the hold time information display part 110 in the receivingsystem 200. If the hold time information packets are received by thereceiving system 200, the hold time information display part 110 informsthe current hold time to the transmitter by controlling a terminal orLED of the transmitter (S50).

[0083] Accordingly, after the transmitter checks the hold time providedfrom the receiving system 200 through the terminal or LED, thetransmitter does not input his or her voice through the microphone 10during the hold time (i.e., he or she does not talk at all).

[0084] In this manner, the compressed voice data generation rate can bemaintained below to the designated channel capacity to the transmitter.

[0085] According to the present invention, as for the voicecommunication over the VoIP system, in order for the compressed voicedata generation rate of the transmission system not to exceed thechannel capacity assigned to the transmitter, the receiving systemcalculates the compressed voice data loss rate and decides the hold timefor temporarily holding the voice input. Therefore, the compressed voicedata loss can be prevented, and the communication quality of the VoIPsystem can be much improved.

[0086] While the invention has been shown and described with referenceto certain preferred embodiments thereof, it will be understood by thoseskilled in the art that various changes in form and details may be madetherein without departing from the spirit and scope of the invention asdefined by the appended claims.

What is claimed is:
 1. A VoIP system for voice communication over the internet, the VoIP system comprising: a receiving system in which the receiving system is equipped with a voice data loss detection part for calculating a compressed voice data packet loss rate per designated time, a protocol header with a sequence number received over the internet, and a hold time decision-making part for deciding a current hold time per designated time using the voice data packet loss rate and generating hold time information packets; and wherein, a transmission system for providing the compressed voice data packets, being equipped with a hold time display part for displaying the hold time based on the hold time information packets that are received over the internet.
 2. The VoIP system of claim 1, wherein the hold time decision-making part has a first allowable value and a second allowable value lower than the first allowable value, and decides the current hold time by comparing the compressed voice data loss rate with a first allowable value, thus if the compressed voice data packet loss rate is larger than the first allowable value, the previous hold time increases by a designated time unit.
 3. The VoIP system of claim 2, wherein the first allowable value is 5%, and the designated time unit is 1 second.
 4. The VoIP system of claim 2, wherein the hold time decision-making part determines a current hold time by decreasing the previous hold time by a designated time unit if the compressed voice data loss rate is lower than a second allowable value.
 5. The VoIP system of claim 4, wherein the second allowable value is 1%, and the designated time unit is 1 second.
 6. The VoIP system of claim 1, wherein the hold time information packet is formatted to comprise an internet protocol header part, a service identifier part A for indicating the hold time information packet, a session ID number part B given to every telephone call, and a hold time part C for indicating a waiting time that commands the user to temporarily stop before staring another sentence during a telephone call.
 7. The VoIP system of claim 6, wherein the service identifier part A is 4 bytes, the session ID number part B is 3 bytes, and the hold time part C is 1 byte.
 8. The VoIP system of claim 7, wherein if the hold time part C is 0000 0000, the hold time is 0 second, if the hold time part C is 0000 0001, the hold time is 1 second, if the hold time part C is 0000 0010, the hold time is 2 seconds, if the hold time part C is 0000 0100, the hold time is 3 seconds, if the hold time part C is 0000 1000, the hold time is 4 second, and if the hold time part C is 0001 0000, the hold time is 5 seconds, and the service identifier part A being 0000 0000 0000 0000 0000 0000 0000
 0000. 9. The VoIP system of claim 1, wherein the designated time unit for calculating the packet loss rate is 30 seconds.
 10. The VoIP system of claim 1, wherein the voice data loss detection part detects that the voice data loss is occurred as much as an increment if the sequence number increases more than 2 units, not increasing 1 unit gradually, while receiving the voice data packets.
 11. A method for preventing data loss in the VoIP system, the method comprising the steps of: transmitting compressed voice data in a form of a packet from the transmission system to the receiving system over the internet; extracting the compressed voice data form the voice data packets received by the receiving system, and calculating a designated time unit of the compressed voice data packet loss rate; deciding a current hold time per designated time unit by comparing the packet loss rate with a first allowable value and a second allowable value that is lower than the first allowable value in the receiving system; and, transmitting the hold time information packet to the transmission system.
 12. The method of claim 11, wherein a step is further included for displaying the hold time based on the hold time information received by the transmission system through a display means.
 13. The method of claim 11, wherein the designated time unit is 30 seconds, the first allowable value is 5%, and the second allowable value is 1%.
 14. The method of claim 11, wherein the total number T of the voice data packets and the voice data packet loss rate L are calculated applying the following algorithm: T=M−N L=(T−D)/T wherein M is the maximum sequence number and N is the minimum sequence number among the voice data packets that are received per designated time unit.
 15. The method of claim 11, wherein the hold time information packet is formatted to comprise an internet protocol header part, a service identifier part A for indicating the hold time information packet, a session ID number part B given to every telephone call, and a hold time part C for indicating a waiting time that commands the user to temporarily stop before staring another sentence during a telephone call.
 16. The method of claim 15, wherein the service identifier part A is 4 bytes, the session ID number part B is 3 bytes, and the hold time part C is 1 byte.
 17. The method of claim 16, wherein if the hold time part C is 0000 0000, the hold time is 0 second, if the hold time part C is 0000 0001, the hold time is 1 second, if the hold time part C is 0000 0010, the hold time is 2 seconds, if the hold time part C is 0000 0100, the hold time is 3 seconds, if the hold time part C is 0000 1000, the hold time is 4 second, and if the hold time part C is 0001 0000, the hold time is 5 seconds, and the service identifier part A being 0000 0000 0000 0000 0000 0000 0000
 0000. 18. The method of claim 11, wherein the procedure, from deciding the current hold time to transmitting the current hold time information in a packet form to the transmission system, comprises the steps of, evaluating whether or not the compressed voice data loss rate is higher than the first allowable value, based on the above evaluation, increasing the previous hold time by 1 time unit if the compressed voice data loss rate is higher than the first allowable value, evaluating whether or not the increased hold time is higher than a designated maximum hold time, based on the above evaluation, designating the increased hold time to be the current hold time if the increased hold time does not exceed the maximum value, and designating the maximum value to be the current hold time if the increased hold time exceeds the maximum value, generating the hold time information packets based on the designated current hold time, and transmitting the hole time information packets to the transmission system over the internet.
 19. The method of claim 18, wherein the designated maximum value is 5 seconds, and the first time unit is 1 second.
 20. The method of claim 11, wherein the procedure, from deciding the current hold time to transmitting the current hold time information in a packet form to the transmission system, comprises the steps of, evaluating whether or not the compressed voice data loss rate is lower than the second allowable value, based on the above evaluation, decreasing the previous hold time by 1 time unit if the compressed voice data loss rate is lower than the second allowable value, evaluating whether or not the decreased hold time is lower than a designated minimum hold time, based on the above evaluation, designating the decreased hold time to be the current hold time if the decreased hold time is not lower than the minimum value, and designating the minimum value to be the current hold time if the decreased hold time is lower than the minimum value, generating the hold time information packets based on the designated current hold time, and transmitting the hold time information packets to the transmission system over the internet.
 21. The method of claim 20, wherein the designated initial minimum value is 0 second, and the first time unit is 1 second. 